annotate mp3lib/decod386.c @ 16682:5f98b89262c4

hqdn3d: 2.5x faster temporal-only, 1.6x faster spatial-only.
author lorenm
date Thu, 06 Oct 2005 06:05:38 +0000
parents 07e7a572bd84
children e7a129082fda
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1 /*
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2 * Modified for use with MPlayer, for details see the CVS changelog at
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3 * http://www.mplayerhq.hu/cgi-bin/cvsweb.cgi/main/
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4 * $Id$
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5 */
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6
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7 /*
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8 * Mpeg Layer-1,2,3 audio decoder
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9 * ------------------------------
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10 * copyright (c) 1995,1996,1997 by Michael Hipp, All rights reserved.
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11 * See also 'README'
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12 *
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13 * slighlty optimized for machines without autoincrement/decrement.
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14 * The performance is highly compiler dependend. Maybe
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15 * the decode.c version for 'normal' processor may be faster
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16 * even for Intel processors.
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17 */
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18
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19
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20 #include "../config.h"
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21
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22 #if 0
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23 /* old WRITE_SAMPLE */
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24 /* is portable */
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25 #define WRITE_SAMPLE(samples,sum,clip) { \
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26 if( (sum) > 32767.0) { *(samples) = 0x7fff; (clip)++; } \
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27 else if( (sum) < -32768.0) { *(samples) = -0x8000; (clip)++; }\
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28 else { *(samples) = sum; } \
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29 }
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30 #else
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31 /* new WRITE_SAMPLE */
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32
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33 /*
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34 * should be the same as the "old WRITE_SAMPLE" macro above, but uses
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35 * some tricks to avoid double->int conversions and floating point compares.
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36 *
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37 * Here's how it works:
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38 * ((((65536.0 * 65536.0 * 16)+(65536.0 * 0.5))* 65536.0)) is
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39 * 0x0010000080000000LL in hex. It computes 0x0010000080000000LL + sum
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40 * as a double IEEE fp value and extracts the low-order 32-bits from the
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41 * IEEE fp representation stored in memory. The 2^56 bit in the constant
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42 * is intended to force the bits of "sum" into the least significant bits
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43 * of the double mantissa. After an integer substraction of 0x80000000
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44 * we have the original double value "sum" converted to an 32-bit int value.
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45 *
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46 * (Is that really faster than the clean and simple old version of the macro?)
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47 */
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48
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49 /*
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50 * On a SPARC cpu, we fetch the low-order 32-bit from the second 32-bit
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51 * word of the double fp value stored in memory. On an x86 cpu, we fetch it
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52 * from the first 32-bit word.
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53 * I'm not sure if the WORDS_BIGENDIAN feature test covers all possible memory
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54 * layouts of double floating point values an all cpu architectures. If
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55 * it doesn't work for you, just enable the "old WRITE_SAMPLE" macro.
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56 */
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57 #if WORDS_BIGENDIAN
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58 #define MANTISSA_OFFSET 1
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59 #else
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60 #define MANTISSA_OFFSET 0
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61 #endif
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62
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63 /* sizeof(int) == 4 */
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64 #define WRITE_SAMPLE(samples,sum,clip) { \
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65 union { double dtemp; int itemp[2]; } u; int v; \
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66 u.dtemp = ((((65536.0 * 65536.0 * 16)+(65536.0 * 0.5))* 65536.0)) + (sum);\
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67 v = u.itemp[MANTISSA_OFFSET] - 0x80000000; \
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68 if( v > 32767) { *(samples) = 0x7fff; (clip)++; } \
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69 else if( v < -32768) { *(samples) = -0x8000; (clip)++; } \
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70 else { *(samples) = v; } \
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71 }
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72 #endif
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73
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74
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75 /*
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76 #define WRITE_SAMPLE(samples,sum,clip) { \
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77 double dtemp; int v; \
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78 dtemp = ((((65536.0 * 65536.0 * 16)+(65536.0 * 0.5))* 65536.0)) + (sum);\
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79 v = ((*(int *)&dtemp) - 0x80000000); \
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80 if( v > 32767) { *(samples) = 0x7fff; (clip)++; } \
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81 else if( v < -32768) { *(samples) = -0x8000; (clip)++; } \
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82 else { *(samples) = v; } \
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83 }
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84 */
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85
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86 static int synth_1to1(real *bandPtr,int channel,unsigned char *out,int *pnt);
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87
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88 static int synth_1to1_mono(real *bandPtr,unsigned char *samples,int *pnt)
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89 {
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90 short samples_tmp[64];
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91 short *tmp1 = samples_tmp;
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92 int i,ret;
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93 int pnt1 = 0;
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94
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95 ret = synth_1to1(bandPtr,0,(unsigned char *) samples_tmp,&pnt1);
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96 samples += *pnt;
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97
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98 for(i=0;i<32;i++) {
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99 *( (short *) samples) = *tmp1;
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100 samples += 2;
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101 tmp1 += 2;
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102 }
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103 *pnt += 64;
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104
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105 return ret;
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106 }
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107
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108
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109 static int synth_1to1_mono2stereo(real *bandPtr,unsigned char *samples,int *pnt)
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110 {
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111 int i,ret;
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112
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113 ret = synth_1to1(bandPtr,0,samples,pnt);
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114 samples = samples + *pnt - 128;
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115
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116 for(i=0;i<32;i++) {
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117 ((short *)samples)[1] = ((short *)samples)[0];
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118 samples+=4;
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119 }
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120
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121 return ret;
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122 }
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123
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124 static synth_func_t synth_func;
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125
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126 #if defined(CAN_COMPILE_X86_ASM)
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127 int synth_1to1_MMX( real *bandPtr,int channel,short * samples)
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128 {
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129 static short buffs[2][2][0x110];
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130 static int bo = 1;
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131 synth_1to1_MMX_s(bandPtr, channel, samples, (short *) buffs, &bo);
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132 return 0;
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133 }
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134 #endif
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135
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136 #ifdef HAVE_ALTIVEC
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137 #define dct64_base(a,b,c) if(gCpuCaps.hasAltiVec) dct64_altivec(a,b,c); else dct64(a,b,c)
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138 #else /* HAVE_ALTIVEC */
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139 #define dct64_base(a,b,c) dct64(a,b,c)
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140 #endif /* HAVE_ALTIVEC */
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141
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142 static int synth_1to1(real *bandPtr,int channel,unsigned char *out,int *pnt)
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143 {
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144 static real buffs[2][2][0x110];
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145 static const int step = 2;
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146 static int bo = 1;
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147 short *samples = (short *) (out + *pnt);
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148 real *b0,(*buf)[0x110];
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149 int clip = 0;
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150 int bo1;
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151
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152 *pnt += 128;
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153
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154 /* optimized for x86 */
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155 #if defined(CAN_COMPILE_X86_ASM)
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156 if ( synth_func )
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157 {
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158 // printf("Calling %p, bandPtr=%p channel=%d samples=%p\n",synth_func,bandPtr,channel,samples);
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159 // FIXME: synth_func() may destroy EBP, don't rely on stack contents!!!
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160 return (*synth_func)( bandPtr,channel,samples);
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161 }
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162 #endif
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163 if(!channel) { /* channel=0 */
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164 bo--;
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165 bo &= 0xf;
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166 buf = buffs[0];
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167 }
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168 else {
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169 samples++;
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170 buf = buffs[1];
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171 }
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172
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173 if(bo & 0x1) {
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174 b0 = buf[0];
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175 bo1 = bo;
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176 dct64_base(buf[1]+((bo+1)&0xf),buf[0]+bo,bandPtr);
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177 }
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178 else {
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179 b0 = buf[1];
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180 bo1 = bo+1;
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181 dct64_base(buf[0]+bo,buf[1]+bo+1,bandPtr);
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182 }
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183
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184 {
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185 register int j;
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186 real *window = mp3lib_decwin + 16 - bo1;
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187
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188 for (j=16;j;j--,b0+=0x10,window+=0x20,samples+=step)
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189 {
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190 real sum;
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191 sum = window[0x0] * b0[0x0];
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192 sum -= window[0x1] * b0[0x1];
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193 sum += window[0x2] * b0[0x2];
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194 sum -= window[0x3] * b0[0x3];
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195 sum += window[0x4] * b0[0x4];
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196 sum -= window[0x5] * b0[0x5];
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197 sum += window[0x6] * b0[0x6];
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198 sum -= window[0x7] * b0[0x7];
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199 sum += window[0x8] * b0[0x8];
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200 sum -= window[0x9] * b0[0x9];
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201 sum += window[0xA] * b0[0xA];
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202 sum -= window[0xB] * b0[0xB];
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203 sum += window[0xC] * b0[0xC];
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204 sum -= window[0xD] * b0[0xD];
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205 sum += window[0xE] * b0[0xE];
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206 sum -= window[0xF] * b0[0xF];
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207
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208 WRITE_SAMPLE(samples,sum,clip);
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209 }
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210
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211 {
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212 real sum;
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213 sum = window[0x0] * b0[0x0];
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214 sum += window[0x2] * b0[0x2];
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215 sum += window[0x4] * b0[0x4];
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216 sum += window[0x6] * b0[0x6];
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217 sum += window[0x8] * b0[0x8];
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218 sum += window[0xA] * b0[0xA];
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219 sum += window[0xC] * b0[0xC];
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220 sum += window[0xE] * b0[0xE];
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221 WRITE_SAMPLE(samples,sum,clip);
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222 b0-=0x10,window-=0x20,samples+=step;
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223 }
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224 window += bo1<<1;
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225
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226 for (j=15;j;j--,b0-=0x10,window-=0x20,samples+=step)
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227 {
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228 real sum;
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229 sum = -window[-0x1] * b0[0x0];
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230 sum -= window[-0x2] * b0[0x1];
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231 sum -= window[-0x3] * b0[0x2];
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232 sum -= window[-0x4] * b0[0x3];
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233 sum -= window[-0x5] * b0[0x4];
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234 sum -= window[-0x6] * b0[0x5];
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235 sum -= window[-0x7] * b0[0x6];
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236 sum -= window[-0x8] * b0[0x7];
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237 sum -= window[-0x9] * b0[0x8];
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238 sum -= window[-0xA] * b0[0x9];
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239 sum -= window[-0xB] * b0[0xA];
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240 sum -= window[-0xC] * b0[0xB];
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241 sum -= window[-0xD] * b0[0xC];
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242 sum -= window[-0xE] * b0[0xD];
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243 sum -= window[-0xF] * b0[0xE];
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244 sum -= window[-0x0] * b0[0xF];
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245
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246 WRITE_SAMPLE(samples,sum,clip);
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247 }
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248 }
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249
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250 return clip;
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251
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252 }
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253
12291
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254 #ifdef USE_FAKE_MONO
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255 static int synth_1to1_l(real *bandPtr,int channel,unsigned char *out,int *pnt)
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256 {
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257 int i,ret;
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258
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259 ret = synth_1to1(bandPtr,channel,out,pnt);
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260 out = out + *pnt - 128;
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261
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262 for(i=0;i<32;i++) {
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263 ((short *)out)[1] = ((short *)out)[0];
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264 out+=4;
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265 }
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266
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267 return ret;
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268 }
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269
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270 static int synth_1to1_r(real *bandPtr,int channel,unsigned char *out,int *pnt)
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271 {
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272 int i,ret;
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273
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274 ret = synth_1to1(bandPtr,channel,out,pnt);
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275 out = out + *pnt - 128;
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276
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277 for(i=0;i<32;i++) {
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278 ((short *)out)[0] = ((short *)out)[1];
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279 out+=4;
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280 }
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281
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282 return ret;
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283 }
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284 #endif